I raised the Dynlog level to 7 and received the following output results on the transfer portion of the log. Notice the Network Out of Order message at the end of the call. Does something need to be configured in Asterisk to support SIP call transfers?<br>
<br>19:36:44:000 DBUG PHONGLUE domU-12- callid=[31] snd played callid=31 status=0 msg="" reason=end-of-data to CT client on fh="::CTCLIENT1" at host=127.0.0.1 proto=SATC<br>19:36:44:000 DBUG PHONGLUE domU-12- snd "played 31 0 \"\" 1\n" to ::CTCLIENT1<br>
19:36:44:001 DBUG VOICEGLU domU-12- rcv ctsrv: "played 31 0 \"\" 1\n"<br>19:36:44:001 DBUG VOICEGLU domU-12- callid=[31] parsed played callid=31 status=0 msg="" reason=end-of-data<br>19:36:44:001 DBUG VOICEGLU domU-12- callid=[31] dpal(): do_prompt_and_listen() called<br>
19:36:44:001 DBUG VOICEGLU domU-12- callid=[31] dpal(): Checking for prompts<br>19:36:44:001 DBUG VOICEGLU domU-12- callid=[31] dpal(): Found prompts to play, playing<br>19:36:44:001 DBUG VOICEGLU domU-12- callid=[31] snd play callid=31 files=("voiceglue/tts/Transferring_call") stopkeys="0123456789*#" to CT server on fh="::CTSRV" at host=localhost<br>
19:36:44:001 DBUG VOICEGLU domU-12- snd "play 31 \"voiceglue/tts/Transferring_call\" 0123456789*#\n" to ::CTSRV<br>19:36:44:001 DBUG PHONGLUE domU-12- rcv ct(::CTCLIENT1): "play 31 \"voiceglue/tts/Transferring_call\" 0123456789*#\n"<br>
19:36:44:002 DBUG PHONGLUE domU-12- callid=[31] parsed play callid=31 files=("voiceglue/tts/Transferring_call") stopkeys="0123456789*#"<br>19:36:44:002 DBUG PHONGLUE domU-12- callid=[31] snd PLAYFILE file="voiceglue/tts/Transferring_call" stopkeys="0123456789*#" to AGI client on fh="::FASTAGI31" at host=127.0.0.1 callid=[31]<br>
19:36:44:002 DBUG PHONGLUE domU-12- snd "STREAM FILE voiceglue/tts/Transferring_call 0123456789*#\n" to ::FASTAGI31<br>19:36:45:567 DBUG PHONGLUE domU-12- rcv agi(::FASTAGI31): "200 result=0 endpos=12537\n"<br>
19:36:45:567 DBUG PHONGLUE domU-12- callid=[31] AGI result=0 values: endpos=12537<br>19:36:45:567 DBUG PHONGLUE domU-12- callid=[31] snd played callid=31 status=0 msg="" reason=end-of-data to CT client on fh="::CTCLIENT1" at host=127.0.0.1 proto=SATC<br>
19:36:45:568 DBUG PHONGLUE domU-12- snd "played 31 0 \"\" 1\n" to ::CTCLIENT1<br>19:36:45:568 DBUG VOICEGLU domU-12- rcv ctsrv: "played 31 0 \"\" 1\n"<br>19:36:45:568 DBUG VOICEGLU domU-12- callid=[31] parsed played callid=31 status=0 msg="" reason=end-of-data<br>
19:36:45:568 DBUG VOICEGLU domU-12- callid=[31] dpal(): do_prompt_and_listen() called<br>19:36:45:568 DBUG VOICEGLU domU-12- callid=[31] dpal(): Sending response to wait<br>19:36:45:569 DBUG VOICEGLU domU-12- callid=[31] dpal() Data Cleared<br>
19:36:45:569 DBUG VOICEGLU domU-12- callid=[31] snd Waited to VXML interpreter on fh="::PERL_VXML_31" at host=localhost callid=[31]<br>19:36:45:569 DBUG VOICEGLU domU-12- snd "Waited\n" to ::PERL_VXML_31<br>
19:36:45:571 DBUG VOICEGLU domU-12- callid=[31] rcv ovxi: "\n"<br>19:36:45:571 INFO VOICEGLU domU-12- callid=[31] had its VXML interpreter thread exit<br>19:36:45:571 DBUG VOICEGLU domU-12- callid=[31] deallocating VXML thread<br>
19:36:45:571 NOTI VOICEGLU domU-12- callid=[31] hanging up because VXML interpreter exited<br>19:36:45:571 DBUG VOICEGLU domU-12- callid=[31] snd hangup to CT server on fh="::CTSRV" at host=localhost<br>19:36:45:571 DBUG VOICEGLU domU-12- snd "hangup 31\n" to ::CTSRV<br>
19:36:45:572 DBUG OPEN_VXI domU-12- callid=[31] rcv vg: Waited<br>19:36:45:572 DBUG OPEN_VXI domU-12- callid=[31] |-1229898864|31|60001|testClient::ChannelThread|NULL result<br>19:36:45:572 DBUG OPEN_VXI domU-12- callid=[31] |-1229898864|31|60001|testClient::ChannelThread|Call Terminated<br>
19:36:45:572 NOTI OPEN_VXI domU-12- callid=[31] Channel 31: Call Terminated<br>19:36:45:572 DBUG OPEN_VXI domU-12- callid=[31] |-1229898864|-1|3000|SBinetDestroyResource|entering: 0x0x8ba4e90 (0x0x8afb5d0)<br>19:36:45:572 DBUG OPEN_VXI domU-12- callid=[31] |-1229898864|-1|3000|SBinetDestroyResource|exiting, returned 0<br>
19:36:45:572 DBUG OPEN_VXI domU-12- callid=[31] snd vg: <br>19:36:45:574 DBUG PHONGLUE domU-12- rcv ct(::CTCLIENT1): "hangup 31\n"<br>19:36:45:574 DBUG PHONGLUE domU-12- callid=[31] parsed hangup<br>19:36:45:574 INFO PHONGLUE domU-12- AGI client filehandle "::FASTAGI31" stopped: hangup requested<br>
19:36:45:574 DBUG PHONGLUE domU-12- callid=[31] snd hungup callid=31 to CT client on fh="::CTCLIENT1" at host=127.0.0.1 proto=SATC<br>19:36:45:574 DBUG PHONGLUE domU-12- snd "hungup 31\n" to ::CTCLIENT1<br>
19:36:45:574 DBUG PHONGLUE domU-12- rcv mgr(::ASTMGR): "Event: Newexten\r\nPrivilege: call,all\r\nChannel: SIP/sip2sip.info-086ee748\r\nContext: phoneglue\r\nExtension: 183\r\nPriority: 3\r\nApplication: Hangup\r\nAppData: \r\nUniqueid: asterisk-1270755393.30\r\n\r\n"<br>
19:36:45:574 DBUG PHONGLUE domU-12- rcv mgr(::ASTMGR): "Event: Hangup\r\nPrivilege: call,all\r\nChannel: SIP/sip2sip.info-086ee748\r\nUniqueid: asterisk-1270755393.30\r\nCause: 38\r\nCause-txt: Network out of order\r\n\r\n"<br>
19:36:45:574 DBUG VOICEGLU domU-12- rcv ctsrv: "hungup 31\n"<br>19:36:45:574 DBUG VOICEGLU domU-12- callid=[31] parsed hungup callid=31<br><br>Thanks,<br>Chris<br><br><div class="gmail_quote">On Thu, Apr 8, 2010 at 10:43 AM, Chris Matthieu <span dir="ltr"><<a href="mailto:chris@getvocal.com">chris@getvocal.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Does Asterisk/VoiceGlue support SIP transfers? I am trying to transfer a SIP call to another SIP address and have not had any success with the following two examples:<br>
<br><?xml version="1.0" encoding="UTF-8"?><br>
<vxml xmlns="<a href="http://www.w3.org/2001/vxml" target="_blank">http://www.w3.org/2001/vxml</a>" xmlns:xsi="<a href="http://www.w3.org/2001/XMLSchema-instance" target="_blank">http://www.w3.org/2001/XMLSchema-instance</a>" xsi:schemaLocation="<a href="http://www.w3.org/2001/vxml" target="_blank">http://www.w3.org/2001/vxml</a> <a href="http://www.w3.org/TR/voicexml20/vxml.xsd" target="_blank">http://www.w3.org/TR/voicexml20/vxml.xsd</a>" version="2.0" application="/root.vxml"><br>
<form id="form1"><br><block><prompt>Transferring call</prompt></block><br><transfer dest="<a href="mailto:sip%3A17476491417@proxy01.sipphone.com" target="_blank">sip:17476491417@proxy01.sipphone.com</a>" /><br>
</form><br></vxml><br><br>OR<br><br><transfer dest="<a href="mailto:17476491417@proxy01.sipphone.com" target="_blank">17476491417@proxy01.sipphone.com</a>" /><br><br>Please advise.<br><br>Thanks,<br>
<font color="#888888">Chris<br><br>
</font></blockquote></div><br>